Diseño de algoritmos de guerra electrónica y radar para su implementación en sistemas de tiempo real

Esta tesis se centra en el estudio y desarrollo de algoritmos de guerra electrónica {electronic warfare, EW) y radar para su implementación en sistemas de tiempo real. La llegada de los sistemas de radio, radar y navegación al terreno militar llevó al desarrollo de tecnologías para combatirlos. Así, el objetivo de los sistemas de guerra electrónica es el control del espectro electomagnético. Una de la funciones de la guerra electrónica es la inteligencia de señales {signals intelligence, SIGINT), cuya labor es detectar, almacenar, analizar, clasificar y localizar la procedencia de todo tipo de señales presentes en el espectro. El subsistema de inteligencia de señales dedicado a las señales radar es la inteligencia electrónica {electronic intelligence, ELINT). Un sistema de tiempo real es aquel cuyo factor de mérito depende tanto del resultado proporcionado como del tiempo en que se da dicho resultado. Los sistemas radar y de guerra electrónica tienen que proporcionar información lo más rápido posible y de forma continua, por lo que pueden encuadrarse dentro de los sistemas de tiempo real. La introducción de restricciones de tiempo real implica un proceso de realimentación entre el diseño del algoritmo y su implementación en plataformas “hardware”. Las restricciones de tiempo real son dos: latencia y área de la implementación. En esta tesis, todos los algoritmos presentados se han implementado en plataformas del tipo field programmable gate array (FPGA), ya que presentan un buen compromiso entre velocidad, coste total, consumo y reconfigurabilidad. La primera parte de la tesis está centrada en el estudio de diferentes subsistemas de un equipo ELINT: detección de señales mediante un detector canalizado, extracción de los parámetros de pulsos radar, clasificación de modulaciones y localization pasiva. La transformada discreta de Fourier {discrete Fourier transform, DFT) es un detector y estimador de frecuencia quasi-óptimo para señales de banda estrecha en presencia de ruido blanco. El desarrollo de algoritmos eficientes para el cálculo de la DFT, conocidos como fast Fourier transform (FFT), han situado a la FFT como el algoritmo más utilizado para la detección de señales de banda estrecha con requisitos de tiempo real. Así, se ha diseñado e implementado un algoritmo de detección y análisis espectral para su implementación en tiempo real. Los parámetros más característicos de un pulso radar son su tiempo de llegada y anchura de pulso. Se ha diseñado e implementado un algoritmo capaz de extraer dichos parámetros. Este algoritmo se puede utilizar con varios propósitos: realizar un reconocimiento genérico del radar que transmite dicha señal, localizar la posición de dicho radar o bien puede utilizarse como la parte de preprocesado de un clasificador automático de modulaciones. La clasificación automática de modulaciones es extremadamente complicada en entornos no cooperativos. Un clasificador automático de modulaciones se divide en dos partes: preprocesado y el algoritmo de clasificación. Los algoritmos de clasificación basados en parámetros representativos calculan diferentes estadísticos de la señal de entrada y la clasifican procesando dichos estadísticos. Los algoritmos de localization pueden dividirse en dos tipos: triangulación y sistemas cuadráticos. En los algoritmos basados en triangulación, la posición se estima mediante la intersección de las rectas proporcionadas por la dirección de llegada de la señal. En cambio, en los sistemas cuadráticos, la posición se estima mediante la intersección de superficies con igual diferencia en el tiempo de llegada (time difference of arrival, TDOA) o diferencia en la frecuencia de llegada (frequency difference of arrival, FDOA). Aunque sólo se ha implementado la estimación del TDOA y FDOA mediante la diferencia de tiempos de llegada y diferencia de frecuencias, se presentan estudios exhaustivos sobre los diferentes algoritmos para la estimación del TDOA, FDOA y localización pasiva mediante TDOA-FDOA. La segunda parte de la tesis está dedicada al diseño e implementación filtros discretos de respuesta finita (finite impulse response, FIR) para dos aplicaciones radar: phased array de banda ancha mediante filtros retardadores (true-time delay, TTD) y la mejora del alcance de un radar sin modificar el “hardware” existente para que la solución sea de bajo coste. La operación de un phased array de banda ancha mediante desfasadores no es factible ya que el retardo temporal no puede aproximarse mediante un desfase. La solución adoptada e implementada consiste en sustituir los desfasadores por filtros digitales con retardo programable. El máximo alcance de un radar depende de la relación señal a ruido promedio en el receptor. La relación señal a ruido depende a su vez de la energía de señal transmitida, potencia multiplicado por la anchura de pulso. Cualquier cambio hardware que se realice conlleva un alto coste. La solución que se propone es utilizar una técnica de compresión de pulsos, consistente en introducir una modulación interna a la señal, desacoplando alcance y resolución. ABSTRACT This thesis is focused on the study and development of electronic warfare (EW) and radar algorithms for real-time implementation. The arrival of radar, radio and navigation systems to the military sphere led to the development of technologies to fight them. Therefore, the objective of EW systems is the control of the electromagnetic spectrum. Signals Intelligence (SIGINT) is one of the EW functions, whose mission is to detect, collect, analyze, classify and locate all kind of electromagnetic emissions. Electronic intelligence (ELINT) is the SIGINT subsystem that is devoted to radar signals. A real-time system is the one whose correctness depends not only on the provided result but also on the time in which this result is obtained. Radar and EW systems must provide information as fast as possible on a continuous basis and they can be defined as real-time systems. The introduction of real-time constraints implies a feedback process between the design of the algorithms and their hardware implementation. Moreover, a real-time constraint consists of two parameters: Latency and area of the implementation. All the algorithms in this thesis have been implemented on field programmable gate array (FPGAs) platforms, presenting a trade-off among performance, cost, power consumption and reconfigurability. The first part of the thesis is related to the study of different key subsystems of an ELINT equipment: Signal detection with channelized receivers, pulse parameter extraction, modulation classification for radar signals and passive location algorithms. The discrete Fourier transform (DFT) is a nearly optimal detector and frequency estimator for narrow-band signals buried in white noise. The introduction of fast algorithms to calculate the DFT, known as FFT, reduces the complexity and the processing time of the DFT computation. These properties have placed the FFT as one the most conventional methods for narrow-band signal detection for real-time applications. An algorithm for real-time spectral analysis for user-defined bandwidth, instantaneous dynamic range and resolution is presented. The most characteristic parameters of a pulsed signal are its time of arrival (TOA) and the pulse width (PW). The estimation of these basic parameters is a fundamental task in an ELINT equipment. A basic pulse parameter extractor (PPE) that is able to estimate all these parameters is designed and implemented. The PPE may be useful to perform a generic radar recognition process, perform an emitter location technique and can be used as the preprocessing part of an automatic modulation classifier (AMC). Modulation classification is a difficult task in a non-cooperative environment. An AMC consists of two parts: Signal preprocessing and the classification algorithm itself. Featurebased algorithms obtain different characteristics or features of the input signals. Once these features are extracted, the classification is carried out by processing these features. A feature based-AMC for pulsed radar signals with real-time requirements is studied, designed and implemented. Emitter passive location techniques can be divided into two classes: Triangulation systems, in which the emitter location is estimated with the intersection of the different lines of bearing created from the estimated directions of arrival, and quadratic position-fixing systems, in which the position is estimated through the intersection of iso-time difference of arrival (TDOA) or iso-frequency difference of arrival (FDOA) quadratic surfaces. Although TDOA and FDOA are only implemented with time of arrival and frequency differences, different algorithms for TDOA, FDOA and position estimation are studied and analyzed. The second part is dedicated to FIR filter design and implementation for two different radar applications: Wideband phased arrays with true-time delay (TTD) filters and the range improvement of an operative radar with no hardware changes to minimize costs. Wideband operation of phased arrays is unfeasible because time delays cannot be approximated by phase shifts. The presented solution is based on the substitution of the phase shifters by FIR discrete delay filters. The maximum range of a radar depends on the averaged signal to noise ratio (SNR) at the receiver. Among other factors, the SNR depends on the transmitted signal energy that is power times pulse width. Any possible hardware change implies high costs. The proposed solution lies in the use of a signal processing technique known as pulse compression, which consists of introducing an internal modulation within the pulse width, decoupling range and resolution.

Acoustic characterization of open cavities with the image source model

El objetivo de esta tesis es investigar las resonancias acústicas de una cavidad abierta tridimensional, de paredes rectas o inclinadas, mediante un método rápido y eficiente en el dominio del tiempo. Este método modela la respuesta temporal en cualquier punto como la convolución de la forma de onda de la fuente con la respuesta impulsiva de la cavidad, la cual se obtiene como una secuencia de impulsos retardados y atenuados procedentes de la fuente real, el primero, y de las fuentes imágenes especulares, los siguientes (Modelo Fuente Imagen, ISM). Además de las componentes directa y reflejadas en las paredes, la respuesta impulsiva también incluye las contribuciones difractadas en los bordes, obtenidas mediante la generación de las componentes difractadas de cada fuente imagen. Las frecuencias de resonancia acústica de la cavidad abierta son extraídas de los picos de la Función de Respuesta en Frecuencia (FRF), obtenida como la transformada de Fourier de la respuesta temporal correspondiente entre una fuente puntual y un punto cualquiera de la cavidad. Las frecuencias de resonancia acústicas estimadas mediante este Método de Fuentes Imagen + difracción en bordes son validadas por comparación con las que proporciona un Modelo de Elementos Finitos (FEM) y con las medidas experimentalmente, con diferencias menores que el 1.6 % y el 2.7 %, respectivamente. A modo de comparación, las frecuencias de resonancia estimadas para la misma cavidad por el método ISM, cuando no se incluye la difracción en los bordes, difieren en un 5.7 % de las obtenidas experimentalmente. ABSTRACT The goal of this thesis is to investigate the acoustic resonances of a three-dimensional open cavity, with parallel and non-parallel walls, by a fast and efficient method in the time domain. This method models the time response in any point as the convolution of the source waveform with the impulse response of the cavity, which, in turn, is obtained as a sequence of attenuated and delayed impulses coming, the first from the real, and the subsequent from the mirror imaged sources (Image Source Model). Besides direct and wall-reflected components, the impulse response includes also edge-diffracted contributions by generating first order diffraction components for each image source. The acoustic resonance frequencies of the open cavity are extracted from the peaks of the Frequency Response Function (FRF), obtained as the Fourier transform of the corresponding time response between a point source and any point in the cavity. The acoustic resonance frequencies estimated by the Image Source Model + edge diffraction are validated by comparison with those provided by a Finite Element Model (FEM) and the ones measured experimentally, differing less than 1.6 % and 2.7 %, respectively. As a comparison, resonance frequencies estimated with the pure Image Source Model differ by 5.7 % from the measured ones.

Implementación de plug-ins para REAPER: módulo de reverberación convolutiva en tiempo real y sistema para la medición de respuestas acústicas en salas

Existen en el mercado numerosas aplicaciones para la generación de reverberación y para la medición de respuestas al impulso acústicas. Sin embargo, éstas son de precios muy elevados y/o no se permite acceder a su código y, mucho menos, distribuir de forma totalmente libre. Además, las herramientas que ofrecen para la medición de respuestas al impulso requieren de un tedioso proceso para la generación de la señal de excitación, su reproducción y grabación y, finalmente, su post-procesado. Este procedimiento puede llevar en ocasiones al usuario a cometer errores debido a la falta de conocimientos técnicos. El propósito de este proyecto es dar solución a algunos de los inconvenientes planteados. Con tal fin se llevó a cabo el desarrollo e implementación de un módulo de reverberación por convolución particionada en tiempo real, haciendo uso de software gratuito y de libre distribución. En concreto, se eligió la estación digital de trabajo (DAW. Digital Audio Worksation) REAPER de la compañía Cockos. Además de incluir las funcionalidades básicas de edición y secuenciación presentes en cualquier DAW, el programa incluye un entorno para la implementación de efectos de audio en lenguaje JS (Jesusonic), y se distribuye con licencias completamente gratuitas y sin limitaciones de uso. Complementariamente, se propone una extensión para REAPER que permite la medición de respuestas al impulso de recintos acústicos de una forma completamente automatizada y amigable para el usuario. Estas respuestas podrán ser almacenadas y posteriormente cargadas en el módulo de reverberación, permitiendo aplicar sobre nuestras pistas de audio la respuesta acústica de cualquier recinto en el que se hayan realizado medidas. La implementación del sistema de medida de respuestas se llevó a cabo empleando la herramienta ReaScript de REAPER, que permite la ejecución de pequeños scripts Python. El programa genera un Barrido Sinusoidal Logarítmico que excita el recinto acústico cuya respuesta se desea medir, grabando la misma en un archivo .wav. Este procedimiento es sencillo, intuitivo y está al alcance de cualquier usuario doméstico, ya que no requiere la utilización de sofisticado instrumental de medida. ABSTRACT. There are numerous applications in the market for the generation of reverb and measurement of acoustic impulse responses. However, they are usually very costly and closed source. In addition, the provided tools for measuring impulse responses require tedious processes for the generation and reproduction of the excitation signal, the recording of the response and its final post-processing. This procedure can sometimes drive the user to make mistakes due to the lack of technical knowledge. The purpose of this project is to solve some of the mentioned problems. To that end we developed and implemented a real-time partitioned convolution reverb module using free open source software. Specifically, the chosen software was the Cockos’ digital audio workstation (DAW) REAPER. In addition to the basic features included in any DAW, such as editing and sequencing, the program includes an environment for implementing audio effects in JS (Jesusonic) language of free distribution and features an unrestricted license. As an extension for REAPER, we propose a fully automated and user-friendly method for measuring rooms’ acoustic impulse responses. These will be stored and then loaded into the reverb module, allowing the user to apply the acoustical response of any room where measurement have been taken to any audio track. The implementation of the impulse response measurement system was done using REAPER’s ReaScript tool that allows the execution of small Python scripts. The program generates a logarithmic sine sweep that excites the room and its response is recorded in a .wav file. This procedure is simple, intuitive and it is accessible to any home user as it does not require the use of sophisticated measuring equipment.

Análisis de la precisión en la medida del tiempo de reverberación y de los parámetros asociados

El trabajo fin de master “Análisis de la precisión en la medida del tiempo de reverberación y de los parámetros asociados” tiene como objetivo primordial la evaluación de los parámetros y métodos utilizados para la obtención de estos, a través del tiempo de reverberación, tanto de forma global, conjunto de todos los métodos, como cada uno de ellos por separado. Un objetivo secundario es la evaluación de la incertidumbre en función del método de medición usado. Para realizarlo, se van a aprovechar las mediciones realizadas para llevar a cabo el proyecto fin de carrera [1], donde se medía el tiempo de reverberación en dos recintos diferentes usando el método del ruido interrumpido y el método de la respuesta impulsiva integrada con señales distintas. Las señales que han sido utilizadas han sido señales impulsivas de explosión de globos, disparo de pistola, claquetas y, a través de procesado digital, señales periódicas pseudoaleatorias MLS y barridos de tonos puros. La evaluación que se realizará a cada parámetro ha sido extraída de la norma UNE 89002 [2], [3]y [4]. Se determinará si existen valores aberrantes tanto por el método de Grubbs como el de Cochran, e interesará conocer la veracidad, precisión, repetibilidad y reproducibilidad de los resultados obtenidos. Los parámetros que han sido estudiados y evaluados son el tiempo de reverberación con caída de 10 dB, (T10), con caída de 15 dB (T15), con caída de 20 dB (T20), con caída de 30 dB (T30), el tiempo de la caída temprana (EDT), el tiempo final (Ts), claridad (C20, C30, C50 y C80) y definición (D50 y D80). Dependiendo de si el parámetro hace referencia al recinto o si varía en función de la relación entre la posición de fuente y micrófono, su estudio estará sujeto a un procedimiento diferente de evaluación. ABSTRACT. The master thesis called “Analysis of the accuracy in measuring the reverberation time and the associated parameters” has as the main aim the assessment of parameters and methods used to obtain these through reverberation time, both working overall, set of all methods, as each of them separately. A secondary objective is to evaluate the uncertainty depending on the measurement method used. To do this, measurements of [1] will be used, where they were carried on in two different spaces using the interrupted noise method and the method of impulse response integrated with several signals. The signals that have been used are impulsive signals such as balloon burst, gunshot, slates and, through digital processing, periodic pseudorandom signal MLS and swept pure tone. The assessment that will be made to each parameter has been extracted from the UNE 89002 [2], [3] and [4]. It will determine whether there are aberrant values both through Grubbs method and Cochran method, to say so, if a value is inconsistent with the rest of the set. In addition, it is interesting to know the truthfulness, accuracy, repeatability and reproducibility of results obtained from the first part of this rule. The parameters that are going to be evaluated are reverberation time with 10 dB decay, (T10), with 15 dB decay (T15), with 20 dB decay (T20), with 30 dB decay (T30), the Early Decay Time (EDT), the final time (Ts), clarity (C20, C30, C50 y C80) and definition (D50 y D80). Depending on whether the parameter refers to the space or if it varies depending on the relationship between source and microphone positions, the study will be related to a different evaluation procedure.

Development of a sensorimotor algorithm able to deal with unforeseen pushes and its implementation based on VHDL

Development of a Sensorimotor Algorithm Able to Deal with Unforeseen Pushes and Its Implementation Based on VHDL is the title of my thesis which concludes my Bachelor Degree in the Escuela Técnica Superior de Ingeniería y Sistemas de Telecomunicación of the Universidad Politécnica de Madrid. It encloses the overall work I did in the Neurorobotics Research Laboratory from the Beuth Hochschule für Technik Berlin during my ERASMUS year in 2015. This thesis is focused on the field of robotics, specifically an electronic circuit called Cognitive Sensorimotor Loop (CSL) and its control algorithm based on VHDL hardware description language. The reason that makes the CSL special resides in its ability to operate a motor both as a sensor and an actuator. This way, it is possible to achieve a balanced position in any of the robot joints (e.g. the robot manages to stand) without needing any conventional sensor. In other words, the back electromotive force (EMF) induced by the motor coils is measured and the control algorithm responds depending on its magnitude. The CSL circuit contains mainly an analog-to-digital converter (ADC) and a driver. The ADC consists on a delta-sigma modulation which generates a series of bits with a certain percentage of 1's and 0's, proportional to the back EMF. The control algorithm, running in a FPGA, processes the bit frame and outputs a signal for the driver. This driver, which has an H bridge topology, gives the motor the ability to rotate in both directions while it's supplied with the power needed. The objective of this thesis is to document the experiments and overall work done on push ignoring contractive sensorimotor algorithms, meaning sensorimotor algorithms that ignore large magnitude forces (compared to gravity) applied in a short time interval on a pendulum system. This main objective is divided in two sub-objectives: (1) developing a system based on parameterized thresholds and (2) developing a system based on a push bypassing filter. System (1) contains a module that outputs a signal which blocks the main Sensorimotor algorithm when a push is detected. This module has several different parameters as inputs e.g. the back EMF increment to consider a force as a push or the time interval between samples. System (2) consists on a low-pass Infinite Impulse Response digital filter. It cuts any frequency considered faster than a certain push oscillation. This filter required an intensive study on how to implement some functions and data types (fixed or floating point data) not supported by standard VHDL packages. Once this was achieved, the next challenge was to simplify the solution as much as possible, without using non-official user made packages. Both systems behaved with a series of interesting advantages and disadvantages for the elaboration of the document. Stability, reaction time, simplicity or computational load are one of the many factors to be studied in the designed systems. RESUMEN. Development of a Sensorimotor Algorithm Able to Deal with Unforeseen Pushes and Its Implementation Based on VHDL es un Proyecto de Fin de Grado (PFG) que concluye mis estudios en la Escuela Técnica Superior de Ingeniería y Sistemas de Telecomunicación de la Universidad Politécnica de Madrid. En él se documenta el trabajo de investigación que realicé en el Neurorobotics Research Laboratory de la Beuth Hochschule für Technik Berlin durante el año 2015 mediante el programa de intercambio ERASMUS. Este PFG se centra en el campo de la robótica y en concreto en un circuito electrónico llamado Cognitive Sensorimotor Loop (CSL) y su algoritmo de control basado en lenguaje de modelado hardware VHDL. La particularidad del CSL reside en que se consigue que un motor haga las veces tanto de sensor como de actuador. De esta manera es posible que las articulaciones de un robot alcancen una posición de equilibrio (p.ej. el robot se coloca erguido) sin la necesidad de sensores en el sentido estricto de la palabra. Es decir, se mide la propia fuerza electromotriz (FEM) inducida sobre el motor y el algoritmo responde de acuerdo a su magnitud. El circuito CSL se compone de un convertidor analógico-digital (ADC) y un driver. El ADC consiste en un modulador sigma-delta, que genera una serie de bits con un porcentaje de 1's y 0's determinado, en proporción a la magnitud de la FEM inducida. El algoritmo de control, que se ejecuta en una FPGA, procesa esta cadena de bits y genera una señal para el driver. El driver, que posee una topología en puente H, provee al motor de la potencia necesaria y le otorga la capacidad de rotar en cualquiera de las dos direcciones. El objetivo de este PFG es documentar los experimentos y en general el trabajo realizado en algoritmos Sensorimotor que puedan ignorar fuerzas de gran magnitud (en comparación con la gravedad) y aplicadas en una corta ventana de tiempo. En otras palabras, ignorar empujones conservando el comportamiento original frente a la gravedad. Para ello se han desarrollado dos sistemas: uno basado en umbrales parametrizados (1) y otro basado en un filtro de corte ajustable (2). El sistema (1) contiene un módulo que, en el caso de detectar un empujón, genera una señal que bloquea el algoritmo Sensorimotor. Este módulo recibe diferentes parámetros como el incremento necesario de la FEM para que se considere un empujón o la ventana de tiempo para que se considere la existencia de un empujón. El sistema (2) consiste en un filtro digital paso-bajo de respuesta infinita que corta cualquier variación que considere un empujón. Para crear este filtro se requirió un estudio sobre como implementar ciertas funciones y tipos de datos (coma fija o flotante) no soportados por las librerías básicas de VHDL. Tras esto, el objetivo fue simplificar al máximo la solución del problema, sin utilizar paquetes de librerías añadidos. En ambos sistemas aparecen una serie de ventajas e inconvenientes de interés para el documento. La estabilidad, el tiempo de reacción, la simplicidad o la carga computacional son algunas de las muchos factores a estudiar en los sistemas diseñados. Para concluir, también han sido documentadas algunas incorporaciones a los sistemas: una interfaz visual en VGA, un módulo que compensa el offset del ADC o la implementación de una batería de faders MIDI entre otras.

VAR models and methods for monetary and health economics

Negli ultimi anni i modelli VAR sono diventati il principale strumento econometrico per verificare se può esistere una relazione tra le variabili e per valutare gli effetti delle politiche economiche. Questa tesi studia tre diversi approcci di identificazione a partire dai modelli VAR in forma ridotta (tra cui periodo di campionamento, set di variabili endogene, termini deterministici). Usiamo nel caso di modelli VAR il test di Causalità di Granger per verificare la capacità di una variabile di prevedere un altra, nel caso di cointegrazione usiamo modelli VECM per stimare congiuntamente i coefficienti di lungo periodo ed i coefficienti di breve periodo e nel caso di piccoli set di dati e problemi di overfitting usiamo modelli VAR bayesiani con funzioni di risposta di impulso e decomposizione della varianza, per analizzare l'effetto degli shock sulle variabili macroeconomiche. A tale scopo, gli studi empirici sono effettuati utilizzando serie storiche di dati specifici e formulando diverse ipotesi. Sono stati utilizzati tre modelli VAR: in primis per studiare le decisioni di politica monetaria e discriminare tra le varie teorie post-keynesiane sulla politica monetaria ed in particolare sulla cosiddetta "regola di solvibilità" (Brancaccio e Fontana 2013, 2015) e regola del GDP nominale in Area Euro (paper 1); secondo per estendere l'evidenza dell'ipotesi di endogeneità della moneta valutando gli effetti della cartolarizzazione delle banche sul meccanismo di trasmissione della politica monetaria negli Stati Uniti (paper 2); terzo per valutare gli effetti dell'invecchiamento sulla spesa sanitaria in Italia in termini di implicazioni di politiche economiche (paper 3). La tesi è introdotta dal capitolo 1 in cui si delinea il contesto, la motivazione e lo scopo di questa ricerca, mentre la struttura e la sintesi, così come i principali risultati, sono descritti nei rimanenti capitoli. Nel capitolo 2 sono esaminati, utilizzando un modello VAR in differenze prime con dati trimestrali della zona Euro, se le decisioni in materia di politica monetaria possono essere interpretate in termini di una "regola di politica monetaria", con specifico riferimento alla cosiddetta "nominal GDP targeting rule" (McCallum 1988 Hall e Mankiw 1994; Woodford 2012). I risultati evidenziano una relazione causale che va dallo scostamento tra i tassi di crescita del PIL nominale e PIL obiettivo alle variazioni dei tassi di interesse di mercato a tre mesi. La stessa analisi non sembra confermare l'esistenza di una relazione causale significativa inversa dalla variazione del tasso di interesse di mercato allo scostamento tra i tassi di crescita del PIL nominale e PIL obiettivo. Risultati simili sono stati ottenuti sostituendo il tasso di interesse di mercato con il tasso di interesse di rifinanziamento della BCE. Questa conferma di una sola delle due direzioni di causalità non supporta un'interpretazione della politica monetaria basata sulla nominal GDP targeting rule e dà adito a dubbi in termini più generali per l'applicabilità della regola di Taylor e tutte le regole convenzionali della politica monetaria per il caso in questione. I risultati appaiono invece essere più in linea con altri approcci possibili, come quelli basati su alcune analisi post-keynesiane e marxiste della teoria monetaria e più in particolare la cosiddetta "regola di solvibilità" (Brancaccio e Fontana 2013, 2015). Queste linee di ricerca contestano la tesi semplicistica che l'ambito della politica monetaria consiste nella stabilizzazione dell'inflazione, del PIL reale o del reddito nominale intorno ad un livello "naturale equilibrio". Piuttosto, essi suggeriscono che le banche centrali in realtà seguono uno scopo più complesso, che è il regolamento del sistema finanziario, con particolare riferimento ai rapporti tra creditori e debitori e la relativa solvibilità delle unità economiche. Il capitolo 3 analizza l’offerta di prestiti considerando l’endogeneità della moneta derivante dall'attività di cartolarizzazione delle banche nel corso del periodo 1999-2012. Anche se gran parte della letteratura indaga sulla endogenità dell'offerta di moneta, questo approccio è stato adottato raramente per indagare la endogeneità della moneta nel breve e lungo termine con uno studio degli Stati Uniti durante le due crisi principali: scoppio della bolla dot-com (1998-1999) e la crisi dei mutui sub-prime (2008-2009). In particolare, si considerano gli effetti dell'innovazione finanziaria sul canale dei prestiti utilizzando la serie dei prestiti aggiustata per la cartolarizzazione al fine di verificare se il sistema bancario americano è stimolato a ricercare fonti più economiche di finanziamento come la cartolarizzazione, in caso di politica monetaria restrittiva (Altunbas et al., 2009). L'analisi si basa sull'aggregato monetario M1 ed M2. Utilizzando modelli VECM, esaminiamo una relazione di lungo periodo tra le variabili in livello e valutiamo gli effetti dell’offerta di moneta analizzando quanto la politica monetaria influisce sulle deviazioni di breve periodo dalla relazione di lungo periodo. I risultati mostrano che la cartolarizzazione influenza l'impatto dei prestiti su M1 ed M2. Ciò implica che l'offerta di moneta è endogena confermando l'approccio strutturalista ed evidenziando che gli agenti economici sono motivati ad aumentare la cartolarizzazione per una preventiva copertura contro shock di politica monetaria. Il capitolo 4 indaga il rapporto tra spesa pro capite sanitaria, PIL pro capite, indice di vecchiaia ed aspettativa di vita in Italia nel periodo 1990-2013, utilizzando i modelli VAR bayesiani e dati annuali estratti dalla banca dati OCSE ed Eurostat. Le funzioni di risposta d'impulso e la scomposizione della varianza evidenziano una relazione positiva: dal PIL pro capite alla spesa pro capite sanitaria, dalla speranza di vita alla spesa sanitaria, e dall'indice di invecchiamento alla spesa pro capite sanitaria. L'impatto dell'invecchiamento sulla spesa sanitaria è più significativo rispetto alle altre variabili. Nel complesso, i nostri risultati suggeriscono che le disabilità strettamente connesse all'invecchiamento possono essere il driver principale della spesa sanitaria nel breve-medio periodo. Una buona gestione della sanità contribuisce a migliorare il benessere del paziente, senza aumentare la spesa sanitaria totale. Tuttavia, le politiche che migliorano lo stato di salute delle persone anziane potrebbe essere necessarie per una più bassa domanda pro capite dei servizi sanitari e sociali.

Equity market linkage and multinational trade accords: The case of NAFTA

This paper examines the impact of multinational trade accords on the degree of stock market linkage using NAFTA as a case study. Besides liberalizing trade among the U.S., Canada and Mexico, NAFTA has also sought to strengthen linkage among stock markets of these countries. If successful, this could lessen the appeal of asset diversification across the North American region and promote a higher degree of market efficiency. We assess the possible impact of NAFTA on market linkage using cross-correlations, multivariate price cointegrating systems, speed of convergence, and generalized variance decompositions of unexpected stock returns. The evidence proves robust and consistently indicates intensified equity market linkage since the NAFTA accord. The results also suggest that interdependent goods markets in the region are a primary reason behind the stronger equity market linkage observed in the post-NAFTA period. (c) 2005 Elsevier Ltd. All rights reserved.

Enhanced detection-guided NLMS estimation of sparse FIR-modeled signal channels

In various signal-channel-estimation problems, the channel being estimated may be well approximated by a discrete finite impulse response (FIR) model with sparsely separated active or nonzero taps. A common approach to estimating such channels involves a discrete normalized least-mean-square (NLMS) adaptive FIR filter, every tap of which is adapted at each sample interval. Such an approach suffers from slow convergence rates and poor tracking when the required FIR filter is "long." Recently, NLMS-based algorithms have been proposed that employ least-squares-based structural detection techniques to exploit possible sparse channel structure and subsequently provide improved estimation performance. However, these algorithms perform poorly when there is a large dynamic range amongst the active taps. In this paper, we propose two modifications to the previous algorithms, which essentially remove this limitation. The modifications also significantly improve the applicability of the detection technique to structurally time varying channels. Importantly, for sparse channels, the computational cost of the newly proposed detection-guided NLMS estimator is only marginally greater than that of the standard NLMS estimator. Simulations demonstrate the favourable performance of the newly proposed algorithm. © 2006 IEEE.

Microphone and loudspeaker arrays for room acoustic measurements

A multichannel spherical speaker array allows, together with a spherical microphones array, the measurement of the MIMO (Multiple Input Multiple Output) acoustic impulse response of an environment capturing meaningful information about propagation of sound between source an receiver. The mathematical framework for extracting arbitrary directivity virtual microphones from real microphones array signals is recalled and the application of the same method to the speakers array to generate arbitrary directivity source is presented. A convenient solutions for the construction and calibration of speakers spherical array for measurement purposes is illustrated. The postprocessing technique developed to compute and visualize acoustic path between source and receiver from measured MIMO impulse response is discussed. Real word results from measurement in a small theater are shown.

Surface acoustic wave devices with low loss and high frequency operation

This thesis describes an industrial research project carried out in collaboration with STC Components, Harlow, Essex. Technical and market trends in the use of surface acoustic wave (SAW) devices are reviewed. As a result, three areas not previously addressed by STC were identified: lower insertion loss designs, higher operating frequencies and improved temperature dependent stability. A review of the temperature performance of alternative lower insertion loss designs,shows that greater use could be made of the on-site quartz growing plant. Data is presented for quartz cuts in the ST-AT range. This data is used to modify the temperature performance of a SAW filter. Several recently identified quartz orientations have been tested. These are SST, LST and X33. Problems associated with each cut are described and devices demonstrated. LST quartz, although sensitive to accuracy of cut, is shown to have an improved temperature coefficient over the normal ST orientation. Results show that its use is restricted due to insertion loss variations with temperature. Effects associated with split-finger transducers on LST-quartz are described. Two low-loss options are studied, coupled resonator filters for very narrow bandwidth applications and single phase unidirectional transducers (SPUDT) for fractional bandwidths up to about 1%. Both designs can be implemented with one quarter wavelength transducer geometries at operating frequencies up to 1GHz. The SPUDT design utilised an existing impulse response model to provide analysis of ladder or rung transducers. A coupled resonator filter at 400MHz is demonstrated with a matched insertion loss of less than 3.5dB and bandwidth of 0.05%. A SPUDT device is designed as a re-timing filter for timing extraction in a long haul PCM transmission system. Filters operating at 565MHz are demonstrated with insertion losses of less than 6dB. This basic SPUDT design is extended to a maximally distributed version and demonstrated at 450MHz with 9.8dB insertion loss.

Approximating differentiable relationships between delay embedded dynamical systems with radical basis functions

This thesis is about the study of relationships between experimental dynamical systems. The basic approach is to fit radial basis function maps between time delay embeddings of manifolds. We have shown that under certain conditions these maps are generically diffeomorphisms, and can be analysed to determine whether or not the manifolds in question are diffeomorphically related to each other. If not, a study of the distribution of errors may provide information about the lack of equivalence between the two. The method has applications wherever two or more sensors are used to measure a single system, or where a single sensor can respond on more than one time scale: their respective time series can be tested to determine whether or not they are coupled, and to what degree. One application which we have explored is the determination of a minimum embedding dimension for dynamical system reconstruction. In this special case the diffeomorphism in question is closely related to the predictor for the time series itself. Linear transformations of delay embedded manifolds can also be shown to have nonlinear inverses under the right conditions, and we have used radial basis functions to approximate these inverse maps in a variety of contexts. This method is particularly useful when the linear transformation corresponds to the delay embedding of a finite impulse response filtered time series. One application of fitting an inverse to this linear map is the detection of periodic orbits in chaotic attractors, using suitably tuned filters. This method has also been used to separate signals with known bandwidths from deterministic noise, by tuning a filter to stop the signal and then recovering the chaos with the nonlinear inverse. The method may have applications to the cancellation of noise generated by mechanical or electrical systems. In the course of this research a sophisticated piece of software has been developed. The program allows the construction of a hierarchy of delay embeddings from scalar and multi-valued time series. The embedded objects can be analysed graphically, and radial basis function maps can be fitted between them asynchronously, in parallel, on a multi-processor machine. In addition to a graphical user interface, the program can be driven by a batch mode command language, incorporating the concept of parallel and sequential instruction groups and enabling complex sequences of experiments to be performed in parallel in a resource-efficient manner.

Ultrafast all-optical integrator based on a fiber Bragg grating:proposal and design

We demonstrate a simple technique for the implementation of an all-optical integrator based on a uniform-period fiber Bragg grating (FBG) in reflection that is designed to present a decreasing exponential impulse response. The proposed FBG integrator is readily feasible and can perform close to ideal integration of few-picosecond and subpicosecond pulses.

Price discovery for Chinese shares cross-listed in multiple markets

In this article we study the relationship between security returns cross-listed on the A share market of China and the H share market at the Stock Exchange of Hong Kong (SEHK). Most of these securities are also cross-listed on other markets. An important feature of this article is that we focus on the multilateral relationships between all cross-listed markets rather than concentrating only on the bi-lateral relationship between A and Hong Kong H shares. Using the impulse response functions and the variance decompositions from a Vector Autoregressive (VAR) process we show that the returns to the A share market are almost exclusively determined by domestic factors. In contrast, we find that the H share market is influenced by both the A share market within China and foreign stock markets elsewhere in the world. Impulse response functions suggest that innovations to the A share market and the Hong Kong H share market are partly transmitted to each other and to stock markets outside China. We show that liquidity has an important role to play in determining the impact that the home market has on cross-listed variance decompositions. © 2012 Copyright Taylor and Francis Group, LLC.

Reach improvement of mode division multiplexed systems using fiber splices

This letter proposes the introduction of discrete modal crosstalk (XT) through fiber splices for the improvement of the distance reach (DR) of mode division multiplexed (MDM) transmission systems over few mode fibers (FMFs). The proposed method increases the DR, reducing the time spread of the FMFs' impulse response. The effectiveness of this method is assessed through simulation considering 3 × 136-Gbit/s MDM-coherently-detected polarization-multiplexed quadrature-phase-shift-keying ultralong haul transmission systems employing inherently low differential mode delay (DMD) FMFs or DMD compensated FMFs. A maximum DR increase factor of 1.9 is obtained for the optimum number of splices per span and optimum splice XT level. © 1989-2012 IEEE.

100-Gb/s point-to-point solutions for long-reach passive optical networks in sparse rural and urban areas

Long reach-passive optical networks (LR-PON) are being proposed as a means of enabling ubiquitous fiber-to-the-home (FTTH) by massive sharing of network resources and therefore reducing per customer costs to affordable levels. In this paper, we analyze the chain solutions for LR-PON deployment in urban and rural areas at 100-Gb/s point-to-point transmission using dual polarization-quaternary phase shift-keying (DP-QPSK) modulation. The numerical analysis shows that with appropriate finite impulse response (FIR) filter designs, 100-Gb/s transmission can be achieved with at least 512 way split and up to 160 km total distance, which is sufficient for many of the optical paths in a practical situation, for point-to-point link from one LR-PON to another LR-PON through the optical switch at the metro nodes and across a core light path through the core network without regeneration.